1. Field of the Invention
This invention relates generally to communications techniques, and more particularly, to systems and methods for increasing the effective information throughput of a transmission medium.
2. Description of Background Art
At one time, the Internet was primarily the domain of educators, scientists, military personnel, and technophiles. Web pages were utilitarian and simplistic by present-day standards. Many offered text-based information, or provided relatively simple graphical interfaces. Although expensive, elaborate equipment was initially required to access the Internet, this hardware was purchased by large corporations or at governmental expense. By and large, the general consuming public did not have the means, inclination, or desire to access the Internet. But this has all changed.
Throughout the years, the Internet has enjoyed an ever-expanding audience. At present, it is something of a rarity to find a residential premises in the United States that does not have Internet access. A modem-equipped personal computer is almost as ubiquitous as a refrigerator, microwave oven, or VCR (video cassette recorder). In the case of refrigerators, widespread usage does not pose an insurmountable technical hurdle, as the public utility company must merely increase its power generation capacity to keep up with the increased demand. However, in the case of the Internet, increased usage poses problems that are not so readily solved. Hundreds of thousands of individuals attempt to access the Internet every day. But they are not satisfied with text-based web pages or simplistic graphical interfaces—they would like to view real-time moving video images, listen to full-bandwidth audio, and download large files which may be many Megabytes in length. For example, a user may wish to download the contents of an entire 650-MB CD.
Given the types of information that Internet users are presently accessing, heavy demands are placed on the bandwidth capacity of the user's Internet connection. 20 Hz–20 KHz stereo audio requires a bandwidth on the order of 176,400 bytes per second for CD-quality audio. Consider the amount of information that must be transmitted if full-motion color video is to be sent in real time. A screen has 525 lines (pursuant to the North American Television Standards Committee, often referred to as NTSC). There are up to 800 pixels (points) on each line. A pixel may be stored as a 24-bit value. To transmit in real time, at least 30 images (frames) should be sent every second. This adds up to a grand total of 302,400,000 bits of information to be transmitted every second.
Telephone lines are required to carry voice information at roughly 2400 bps. A single ISDN line can carry information at 64 Kbps. So, consider how many telephone lines would be required to carry the amount of information in a full-motion video transmission. In most situations, and certainly for the typical residential consumer, the use of multiple telephone lines would be impractical.
In view of the aforementioned limitations, various video compression schemes have been developed. A first level of compression uses a single 64-kbps ISDN line to provide a grainy, choppy, and “jerky” image which could be used for non-critical videoconferencing applications. However, many users consider these images to be of unacceptably low quality.
Moving on to the next level, group video conferencing systems compress the video signal to a value within the range of approximately 128 Kbps to 2 Mbps. By way of example, 384 Kbps provides reasonable picture quality for use in many educational environments. Digital video broadcasting uses rates in the range of 2–6 Mbps. Image quality is greatly increased over the previous compression levels, but a very “broadband” connection is required. The recently-promulgated HDTV standards consume even more bandwidth, as they require rates of 25–45 Mbps.
With the foregoing video data rates in mind, consider that most residential users access the Internet over a conventional subscriber loop to the local telephone company central office. Using state-of-the-art modem technology, this subscriber loop provides a bandwidth no greater than 56 Kilobits per second and, in many cases, a lot less.
Although expanded-bandwidth solutions exist on paper, practical implementations of solutions have not yet been realized. As a practical matter, when a residential customer orders a “broadband” service such as ISDN (Integrated Services Digital Network) or DSL (Digital Subscriber Line) from the local telephone company, it is often difficult or impossible to obtain adequate performance. Even if the various installation obstacles are eventually overcome, the monthly fees associated with an ISDN or DSL connection are cost-prohibitive for many customers.
Cable modems represent another broadband alternative for Internet access. Nonetheless, telephone service is generally regarded as a basic necessity, present in virtually every household throughout the United States, whereas cable service is considered to be more of a luxury or non-essential item. Moreover, many residential premises have active and working telephone jacks installed throughout, whereas activated cable jacks may not be present in the vicinity of the computer room. Even if the customer pays the cable company to install an additional jack, this still restricts Internet access to one or two locations. Moreover, as in the case of DSL and ISDN, the subscription charge for Internet access via a cable modem is cost-prohibitive for many customers.
What is needed is an improved method for accessing the Internet over an ubiquitous, inexpensive, widely-available communications link. Such a method should not require the installation of additional wiring to a residential premises and, ideally, should operate over presently-existing communication paths. One possible candidate for such a communications path is the existing public switched telephone network (PSTN).
PSTN: As the public switched telephone network evolved, copper wire pairs were utilized in a manner such that a single pair would carry only one voice message at a time. Communications companies soon realized that, in order to enlarge their message-carrying capacity, they would have to devise ways to transmit several messages simultaneously over a single wire pair, because the cost of installing additional wires to accommodate increased demand was high. Companies would have a competitive advantage if they could reduce costs by putting more and more information over a single wire pair. Over time, discoveries in transmission techniques enabled more than one message to be transmitted per wire pair, thereby paving the way for the telephone industry to become a viable commercial enterprise.
The challenge of maximizing effective bandwidth and increasing line capacity existed from the very beginning of telecommunications technology, and is still with us today. Presently, telecommunications networks are the primary mechanism for conveying voice and data traffic from one location to another. But existing telecommunication networks cannot handle the ever-increasing demand for transmission capacity. Rising population, lower telephone rates, and increased data traffic over the Internet, all underscore the need to increase network capacity. As more and more bandwidth becomes available, higher bandwidth applications are quickly developed, such as higher-resolution web pages and video-on-demand, which once again heightens the demand for increased bandwidth and/or improved information throughput.
One way to satisfy an increasing demand for bandwidth is by installing additional transmission lines or by placing additional satellites in the sky. Both solutions are expensive and dictate substantial investments. Yet, even satellite solutions have limitations, for there is only a limited number of satellites that can be placed in geostationary orbit in the Clarke belt. The Clarke belt is the only location where satellites, when viewed from the Earth's surface, remain substantially stationary, thereby permitting the use of fix-mounted dish antennas. Moreover, owing to the fact that satellite communications operates at the speed of light, the substantial distance between satellites and earth stations introduces a perceptible propagation delay into the signal path. This delay can be disturbing, annoying, and frustrating in the case of real-time interactive applications. On the other hand, terrestrial-based wireless systems operate over the public radio spectrum, which, by its very nature, is a limited resource. Bandwidth utilization and compression methods maybe employed to expand the capacity of wireless systems, but these methods are not sufficient to meet demand in heavily-populated areas. To remain competitive, network service providers must endeavor to preserve the functionality of their existing networks, yet still be able to accommodate the increasing bandwidth demand to handle voice, data, and video transmission.
In conventional analog transmission, acoustical energy from a speaker's voice vibrates a diaphragm or crystal in a microphone. The crystal or diaphragm is used to transform these mechanical vibrations into an electrical signal. The amplitude of this electrical signal varies in a manner analagous to the acoustical vibrations of the speaker's voice. This electrical signal can be amplified and transmitted over a wire pair to a receiver at a remote location. At the receiver, the electrical signal is used to energize an electromagnet, actuating a diaphragm in proximity to the magnet, whereby the diaphragm vibrates to reproduce the original voice. Digital transmission adds several steps to this transformation, starting with an electrical signal from a microphone. This signal has an amplitude which could vary thousands of times per second. These measurements are encoded as voltage or amplitude levels which represent numbers. In the case of binary encoding schemes, the numbers consist of “0's” and “1's”.
Unlike analog transmission which conveys audio information as a continuous waveform, in digital transmission, numbers are transmitted in representational encoding schemes. Digits or bits may be transmitted singly, as discrete, on-off or zero/non-zero current pulses, or in groups as simultaneous pulses at different frequencies. At the receiving end, the bit stream is interpreted and the numbers reconstituted to modulate a current which drives a speaker. This method is “digital” because it entails conversion of an analog signal to numbers, and the transmission of digits in symbolic form.
Compression: There are several known methods which provide for the transmission of information while reducing the overall bandwidth requirements. The most widely employed compression method uses mathematical algorithms and dictionary tables to reduce the number of digits needed to represent a given amount of information. As a consequence of the reduced number of digits that need to be transmitted, bandwidth requirements are correspondingly reduced. In practice, compression may be achieved by building a predictive model of a signal waveform, removing unnecessary elements, and reconstructing the waveform from the remaining elements.
When converting an analog signal into digital form, it is necessary for the digitized signal to contain sufficient information so as to enable a subsequent reconstruction of the analog signal. In order to properly reconstruct the analog signal, one must implement at least twice as many measurements (samples per second), as the highest frequency component in the signal. This requirement is oftentimes referred to as the Nyquist Criterion. The human voice generates sound frequencies in an approximate range of 20 to 4,000 Hz. Hence, a digital voice circuit, accepting an input in the range of 0–4,000 Hz, must sample this signal 8,000 times per second. In practice, the PSTN represents each sample using 7 bits of data plus a sign bit, for a total of 8 bits. A single voice circuit, referred to as DSO, “digital signal level zero”, carries 64,000 (8,000×8) bits of data.
Compression methods are based upon reducing the number of bits required to convey a human voice or other data transmission. Currently-utilized compression algorithms can produce acceptable voice quality using less than 64 kbs by eliminating frequencies not necessary for voice intelligibility, particularly those below 300 Hz and those above 3,300 Hz, and possibly by emphasizing frequencies in the 1,000-Hz range that carry most of the voice energy. Unfortunately, some compression methods are carefully tailored for voice transmission applications, and tend to drop an excessive amount of information in the case of data or other non-voice signals. These compression methods cause problems when utilized in conjunction with high-speed tonal data transmission schemes employed by modems and faxes. In any event, currently-employed compression algorithms and equipment are able to transmit acceptable voice quality with a compression ratio of 8:1, using 8,000 bps per channel.
Using the foregoing compression methods, one channel can convey eight voice conversations or eight fax transmission over a line that originally was able to carry only one voice conversation. Higher compression methods which transmit voice and data over a circuit using less than 8,000 bps, suffer from increasing degradation of voice quality and “loss,” whereby at the receiving end of the line the voice (in its original form) is not heard clearly and distinctly. Although new methods and algorithms may be employed to allow for clear voice transmission using less than 8,000 bps, there are appreciable limitations to these methods. All compression methods using algorithms suffer from greater and greater “loss” as compression ratios increase. Fax and video transmission are more sensitive to bandwidth degradations than voice and, hence, are more limited in their acceptable compression ratios.
While the main advantage of digital compression is that it increases network efficiency, in some cases, it can reduce efficiency. For example, if the amount of time required by a computer to compress and decompress data is relatively lengthy, this can reduce efficiency. Multiplexing: One of the most widely-utilized data transmission protocols is known as “T1”. T1 uses a form of multiplexing in which 24 voice or data channels are multiplexed over a four-wire cable (2 wires for transmit, and two wires for receive). Pursuant to the Nyquist Criterion, a voice channel must be sampled at a rate of approximately 8 KHz so as to permit the rendering of a clear representation of the sampled signal. In other words, one 8-bit sample must be taken every 125 microseconds. Since 24 individual channels must be read for each frame, in addition to a framing bit, the system must transmit 193 bits in 125 microseconds. At this rate, T1 must send or receive data at (193/125×10−6), or 1,544,000 bits per second. Therefore, the total bandwidth capacity of T1 is 1.544 Mbps. Compression methods are used in conjunction with T1 and other transmission protocols to maximize bandwidth. Common compression systems, using a ratio of 8:1, can carry 192 simultaneous voice or data channels (24×8) over a T1 line. Conversations or digital information carried on each of a plurality of T1 lines or channels is rendered unique, and is then transmitted with other T1 channels over a common transmission medium.
Another technique, FDM (Frequency Division Multiplexing), has been employed by phone companies to render each of a plurality of voice channels unique. These voice channels are then carried over a single transmission medium, which is typically a twisted wire pair. Pursuant to one illustrative implementation of FDM, each of a respective 24 voice and/or data channels are assigned to a corresponding frequency band. For example, line 1 is assigned to a frequency band of 0 Hz–4,000 Hz, line 2 is assigned to a frequency band of 4,000 Hz–8,000 Hz band, and so on. This method is best suited for analog signals which are subject to degradation and noise interference.
Other illustrative multiplexing techniques are Time Division Multiplexing (TDM) and Statistical Multiplexing (STDM), often called “packet switching.” Pursuant to TDM, each of 24 channels (or lines) are rendered distinct by assigning each channel to a particular, non-overlapping time slot. Frames of 24 time slots are transmitted, in which Channel 1 is allocated the first time slot in the frame, Channel 2 is allocated the second time slot, and so on. STDM works in a similar manner to TDM, assigning channels on the basis of time division. But STDM takes advantage of statistical fluctuations, and instead of automatically assigning each channel to a time slot, STDM assigns only active channels to time slots. Hence, instead of transmitting channels in sequential order (1, 2, 3, 4, 5, 6) as in TDM, STDM only assigns time slots to channels that are being used, e.g., 1, 6, 3, 5, 6, 5, 3, etc. In general, STDM provides more efficient bandwidth utilization than TDM.